1. Technical Field
The present invention relates generally to speech encoding and decoding in voice communication systems; and, more particularly, it relates to various techniques used with code-excited linear prediction coding to obtain high quality speech reproduction through a limited bit rate communication channel.
2. Related Art
Signal modeling and parameter estimation play significant roles in communicating voice information with limited bandwidth constraints. To model basic speech sounds, speech signals are sampled as a discrete waveform to be digitally processed. In one type of signal coding technique called LPC (linear predictive coding), the signal value at any particular time index is modeled as a linear function of previous values. A subsequent signal is thus linearly predictable according to an earlier value. As a result, efficient signal representations can be determined by estimating and applying certain prediction parameters to represent the signal.
Applying LPC techniques, a conventional source encoder operates on speech signals to extract modeling and parameter information for communication to a conventional source decoder via a communication channel. Once received, the decoder attempts to reconstruct a counterpart signal for playback that sounds to a human ear like the original speech.
A certain amount of communication channel bandwidth is required to communicate the modeling and parameter information to the decoder. In embodiments, for example where the channel bandwidth is shared and real-time reconstruction is necessary, a reduction in the required bandwidth proves beneficial. However, using conventional modeling techniques, the quality requirements in the reproduced speech limit the reduction of such bandwidth below certain levels.
Speech encoding becomes increasingly more difficult as data transmission bit rates decrease. In the absence of embedded intelligence to select an optimal encoding mode or scheme, many speech encoders do not maximize their inherent computational capacity in response to varying operating conditions. Particularly within data transmission systems that operate at varying bit rates, the inability to adapt to a particular encoding scheme based upon the available transmission bit rate at a given time results in an inefficient use of the encoder's resources.
Additionally, the inability to determine the optimal encoding mode for a given speech signal at a given bit rate also contributes to inefficient resource allocation. For a given speech signal and available bit rate, the ability to adaptively select an optimal coding scheme at a given bit rate would provide more efficient use of an encoder processing circuit. Moreover, the inability to select the optimal encoding mode for a given signal after identifying the computational resources required by the various available encoding modes often results in over-dedicating computational resources of a speech encoding system.
Further limitations and disadvantages of conventional systems will become apparent to one of skill in the art after reviewing the remainder of the present application with reference to the drawings.